Amplifier current consumption control

ABSTRACT

The audio amplifier includes a variable gain amplifier receiving the input audio signal and providing the output signal, whereby the output signal corresponds to the input signal amplified by a limiter gain. The audio amplifier further includes a limiter gain calculation unit, thus the input signal is amplified by the limiter gain. A control unit receives a signal representative of the input signal and is configured to estimate, based on a mathematical model, the input current or the total output current of the audio amplifier thus providing an estimated current signal corresponding to (and resulting from) the output signal, whereby the limiter gain calculation unit is configured to calculate, dependent on the estimation, the limiter gain such that the actual input current or the total output current of the audio amplifier does not exceed a threshold current value.

CLAIM OF PRIORITY

This patent application claims priority from EP Application No. 10 196153.0 filed Dec. 21, 2010, which is hereby incorporated by reference.

FIELD OF TECHNOLOGY

The present invention relates to audio amplifiers, in particular to alimiter for controlling the current consumption of an audio amplifier.

RELATED ART

In mobile applications, in particular in automotive applications, powerconsumption of audio amplifiers is often an issue that affects the powersupply of the overall mobile entity.

For example, in automobiles the on board power supply usually provides anominal supply voltage of 12 V that results in a significantly highinput current consumption of electrical loads such as audio amplifierswhose nominal power consumption may be 1000 W or even more. A higher onboard supply voltage (e.g., 42 V), would alleviate the problem andreduce the input current consumption and has been in discussion foryears but, it could not yet be established as standard in automotiveapplications. As a consequence, the supply lines of electrical loadssuch as audio amplifiers have to have significantly large cross sections(or diameters) in order to reduce their ohmic resistance and preventoverheating or even melting of the supply lines. Thick supply linesincrease weight and fuel consumption of the vehicle, both beingundesired consequences.

In order to avoid an over-current in the supply lines a strict maximumcurrent consumption is specified for each electrical load whereby themaximum acceptable current may depend on the total number of activeloads in the vehicle, the battery charge condition, etc. Otherinfluencing factors may be relevant. As far as audio equipment isconcerned suppliers have to comply with requirements related to themaximum output current of the amplifier. As the input currentconsumption of an audio amplifier is a direct result of the respectiveaudio channel output currents current limits may be either specified forthe input current or for the output current.

The actual current consumption of audio amplifiers largely depends onuser settings (e.g., bass, volume, etc.) as well as on the reproducedaudio signal (e.g., voice signal, music with dominating bass, etc).Thus, there is a need for an audio amplifier capable to monitor andcontrol its input current consumption and/or its output currents.

SUMMARY OF THE INVENTION

An audio amplifier includes a variable gain amplifier receiving theinput audio signal and providing the output signal, whereby the outputsignal corresponds to the input signal amplified by a limiter gain. Theaudio amplifier also includes a limiter gain calculation unit, thus theinput signal is amplified by the limiter gain. A control unit receives asignal representative of the input signal and is configured to estimate,based on a mathematical model, the input current or the total outputcurrent of the audio amplifier thus providing an estimated currentsignal corresponding to (and resulting from) the output signal, wherebythe limiter gain calculation unit calculates, dependent on theestimation, the limiter gain such that the actual input current or thetotal output current of the audio amplifier does not exceed a thresholdcurrent value.

These and other objects, features and advantages of the presentinvention will become apparent in the detailed description of the bestmode embodiment thereof, as illustrated in the accompanying drawings. Inthe figures, like reference numerals designate corresponding parts.

DESCRIPTION OF THE DRAWINGS

The invention can be better understood with reference to the followingdrawings and description. The components in the figures are notnecessarily to scale, instead emphasis being placed upon illustratingthe principles of the invention. Moreover, in the figures, likereference numerals designate corresponding parts. In the drawings:

FIG. 1 is a block diagram illustration of an audio system whereby onlyone output channel is shown for the sake of simplicity;

FIG. 2 is a block diagram of an alternative embodiment audio system;

FIG. 3 illustrates the example of FIG. 2 in more detail;

FIG. 4 illustrates the example of FIG. 3 in more detail;

FIG. 5 illustrates one example of the adaptive filter used in theexamples of FIGS. 3 and 4;

FIG. 6 illustrates an alternative to the example illustrated in FIG. 5;

FIG. 7 illustrates the combining of the output signals to a provide asum signal as shown in FIG. 5 further considering the loudspeaker'simpedances; and

FIG. 8 illustrates the example of FIG. 4 using a frequency dependentlimiter gain.

DETAILED DESCRIPTION OF THE INVENTION

The making and using of the presently preferred embodiments arediscussed in detail below. It should be appreciated, however, that thepresent invention provides many applicable inventive concepts that canbe embodied in a wide variety of specific contexts. The specificembodiments discussed are merely illustrative of specific ways to makeand use the invention, and do not limit the scope of the invention.

It has to be noted that in the further discussion (audio) signals areillustrated as discrete time signals. A signal a[n] illustrates ageneral discrete time signal with a time index n denoting the discretesampling time. Further it has to be noted that only these components ofthe depicted systems which are relevant for the further discussion areincluded in the figures. Analog-to-digital converters, digital-to-analogconverters, power amplifiers and other components which may be necessaryfor signal transmission and audio signal reproduction are not shown forthe sake of simplicity.

FIG. 1 is a block diagram illustration of an audio system in accordancewith one example of the invention whereby only one output channel isshown for the sake of simplicity. However, the principle illustrated inthe example of FIG. 1 can readily be generalized to multiple audiochannels. An input signal x[n] (associated with a certain audio channel)is generated by a signal source 10 and may be pre-processed by a signalprocessing unit 20. That is, the actual input signal x[n] is provided atan output of the signal processing unit 20 for further processing.

The signal processing unit 20 typically performs various signalprocessing tasks such as the equalization of the audio signal, surroundsound processing or the like. However, the details of the pre-processingare not essential for the further discussion.

The input signal x[n] is amplified by a variable gain referred to as“limiter gain” Gum[n] whose value may be updated each sampling interval.The resulting output signal y[n] may thus expressed asy[n]=x[n]·G_(LIM)[n]. The output signal y[n] is forwarded to theloudspeaker 40, via an D/A converter and a power amplifier stage, whereit is converted in a corresponding acoustic audio signal.

The limiter gain G_(LIM)[n] is calculated by a limiter gain calculationunit 32 from a predicted current signal i_(IN)′ representative of theinput current consumption (i.e., predicted total input currentconsumption of the amplifier or, alternatively, predicted total outputcurrent provided by the amplifier output stage) and from a currentthreshold i_(MAX) defining the desired maximum current (i.e., inputcurrent consumption or, alternatively, maximum output current). Thelimiter gain G_(LIM)[n] is calculated such that the actual input currenti (or output current) will not exceed the current threshold i_(MAX).However, a short transient over-current might be acceptable depending onthe actual application. The variable gain amplifier 31 and the limitergain calculation unit 32 may be together regarded as limiter 30. Itshould be noted that in the following description, the input currenti_(IN) is predicted and compared to a respective current thresholddefining a maximum input current consumption. As already mentionedabove, a corresponding current threshold may be defined for the totaloutput current of the amplifier which would require to predict the totaloutput current. In the following description the input currentconsumption i_(IN)′ is predicted, however, it will be understood thatsuch input current estimate i_(IN)′ may easily be converted into acorresponding output current estimate as the total output current andthe input current consumption are approximately proportional inpractical audio amplifier implementations.

When predicting a future input current i_(IN)′ value (resulting from theactually present input signal x[n]) a nominal limiter gain value (e.g.,G_(LIM)=1) is assumed. Predicting an over-current i_(IN)′>i_(MAX) allowsfor reducing the limiter gain G_(LIM)[n] (e.g., to values lower thanunity) before the actual input current i actually reaches the predictedvalue i . Alternatively, when a future output current value is to bepredicted, the actual prediction may be performed using input currentinformation resulting in an estimate for the input current that can beconverted to an estimate for the output current.

For the further discussion one aspect is the calculation (i.e., theestimation, also referred to as prediction) of an input currenti_(IN)′[n+1] (or alternatively an output current) resulting from a giveninput signal x[n]. Such prediction makes use of the actual input oroutput current information provided by one or more current sensors suchas by the input current consumption sensor 50 illustrated in FIG. 1. Inthe example of FIG. 1 this calculation is performed by a control unit 33that includes a mathematical model describing the relationship betweeninfluencing variables such as the actual supply voltage u_(IN) of theaudio amplifier, the audio input signal x[n] to be reproduced, thenumber of active loudspeakers, the temperature of the amplifier outputstage, etc. and the resulting input current i_(IN) (or output current).Various other influencing variables might be considered, too. Forenabling an ongoing estimation (prediction) a current sensor 50 may beprovided configured to supply a signal representing the actual inputcurrent i_(IN)[n] to the control unit 33.

FIGS. 2 to 4 illustrate the principle and the details of one possibleway of estimating the impending input current i_(IN)′[n+1] from thecurrent values of the input signal x[n], the (amplified) output signaly[n], and a measured input current sensor signal representing thecurrent input current signal i_(IN)[n] (e.g., stemming from currentconsumption sensor 50). As mentioned above, the corresponding outputcurrent may be estimated as an alternative. FIG. 2 illustrates thecontrol unit 33 receiving the above mentioned signals x[n], y[n], andi_(IN)[n]. Further, FIG. 2 illustrates the transfer function H(z)describing the transfer characteristics from the output signal y[n](output by the loudspeaker 40 via a power amplifier output stage) to theinput current i_(IN)[n] that primarily depends on the output signal y[n]whereby a signal representative of the input current i_(IN)[n] may besensed by the current sensor 50. The mentioned transfer characteristicsmay be estimated by the control unit 33 during operation of the audioamplifier and the corresponding estimated transfer function H′(z) may beused to calculate, from the input signal x[n], an estimation for theimpending input current i_(IN)′[n+1]. This is possible because H′(z) hasa low pass characteristics and always includes a delay, further denotedas delay time t_(H) (e.g., the delay between a peak in signal y[n] and acorresponding peak in the input current i_(IN)[n]). At this point itshould be remembered that the system H(z) is regarded as discrete timesystem in the present discussion as a digital implementation of thecontrol unit 33 seems to be advantageous. However, at least partiallyanalog implementations might be applicable as well.

FIG. 3 illustrates, as one example of the present invention, a usefulimplementation for estimating the transfer characteristics of the systemH(z). An adaptive filter 331 may be implemented in the control unit(e.g., as an adaptive FIR filter) which receives, as input, the outputsignal y[n] and, as reference, the measured input current i_(IN)[n]. Asthe input current i_(IN)[n] is always positive (regardless of whetherthe output signal y[n] has positive or negative values) the outputsignal y[n] may be subject to a positive definite transformation beforebeing supplied to the adaptive filter 331. For example, the squaredoutput signal y[n]² or the absolute signal values |y[n]l may be suppliedto the adaptive filter 331. The transfer function H(z) usuallyrepresents a simple low pass characteristics, as low frequency soundrequires a high sound pressure level so as to be acousticallyperceivable. The adaptive filter H′(z) (filter 331) usually does notchange its coefficients quickly, as the transfer characteristics of theoutput signal y[n] to the input current i_(IN)[n] (or alternatively tothe output current) usually changes only gradually. The actual currentestimate i_(IN)′[+1] is calculated by filtering the (optionally squared)input signal x[n] using a filter 332 which is a (optionally scaled byfactor G_(LIM)[n] to avoid a continuous recalibration of h_(k)[n]) copyof the filter coefficients provided by the adaptive filter 331. In theexample of FIG. 3 the filter 332 implements the same filter coefficientsh_(k)[n] as the adaptive filter 331, and thus the current estimate atthe output of filter 332 is scaled by a scaling factor equal to thecurrent limiter gain G_(LIM)[n]. However, as the current limiter gain is“known” by the limiter gain calculation unit, such scaling can beconsidered in the calculation of the updated limiter gain G_(LIM)[n+1].

As illustrated in FIG. 4, the limiter 30 (composed of, inter alia,variable gain amplifier 31 and limiter gain calculation unit 32) mayfurther include a delay t_(LIM) that may be lumped into the delay unit34 for the purpose of mathematical modeling. The delay t_(LIM) istypically greater than one millisecond for feed forward limiters inpractical implementations. This delay may adversely affect theadaptation behavior of the adaptive filter 331 as the signal y[n]reaches the filter 331 after the resulting current signal i_(IN)[n]. Toavoid this, a delay line 334 (delay Δt≧t_(LIM)-t_(H)) has to be includedbetween the current sensor 50 and the adaptive filter 331 so as to delaythe current sense signal i_(IN)[n] received by the adaptive filter. Therespective delay should be at least as large as the difference betweenthe limiter delay t_(LIM) and the dead time t_(H) of system H(z).

It may be worth noting that the output signal y[n] should be tapped soas to be supplied to the adaptive filter (after having been squared, seeFIG. 4). In theory, also the input signal x[n] might be tapped insteadfor the same purpose. In the latter case, however, the resultingadaptive filter would be permanently resealed in response to a change ofthe limiter gain G_(um)[n] whereas, once having converged, thecoefficients of the adaptive filter 331 remain almost constant whentaking the signal y[n] as input. The current limiter gain valueG_(LIM)[n] might instead be considered in the gain calculation unit 32.

A real world audio system typically has more than one output signaly[n], typically at least two. In the following the output signal of eachchannel is labeled with the index CH1, CH2, etc., that is y_(CH1)[n],y_(CH2)[n],. . . , y_(CHN)[n] . Each of the output signals y_(CH1)[n],y_(CH2)[n], . . . , y_(CHN)[n] contributes to the total currentconsumption i_(IN)[n] of the audio amplifier. FIG. 5 illustrates theestimation of the predicted input current signal i_(IN)′[n] in amulti-channel case. In order to account for the contribution of eachchannel each output signal is squared (see unit 333 in FIGS. 4 and 5)and the squared signals are added so as to form a sum signal y_(SUM)[n]which represents the superposition of the output signals of all outputchannels. Instead of squaring the signals their absolute values may betaken as already mentioned above. The sum signal y_(SUM)[n] is processedessentially in the same way as the signal y[n] in the example of FIGS. 3and 4. The sum signals y_(SUM[n]) ^(and x) _(SUM)[n] may represent aweighted sum. Especially, if the loudspeakers connected to the cannelsdo not have the same impedance the individual signals y_(CH1)[n],y_(CH2)[n], . . . , y_(CHN)[n] (or x_(CH1)[n], x_(CH1)[n], x_(CHN)[n])have to be weighted in accordance with the corresponding loudspeaker'simpedance before calculating the respective sum signal. The details ofthe weighting is explained further below with reference to FIG. 7.

The signal y_(SUM)[n] is supplied to the adaptive filter 331 which iscomposed by a FIR filter unit and an adaptation unit, whereby theadaptation unit iteratively calculates updated filter coefficientsh_(k)′[n] that represent the filter transfer function H′(z). For thispurpose the adaptation unit receives the signal y_(SUM)[n] and an errorsignal e[n] which represents the difference between the output of theadaptive filter (i.e., the estimation) and the actual current signali_(IN)[n]. The iterative calculation is done in accordance with knownadaptation algorithms such as, for example, a Least-Mean-Squaresalgorithms (LMS algorithms). A copy of the adaptive filter coefficientsh_(k)[n] is used in the filter unit 332 for filtering a sum signalx_(SUM)[n] representing a superposition of the (e.g., squared) inputsignals x[n] of all channels. The delay unit 334 shown in FIG. 5 isoptional and may be provided to account for a dead time in the transfercharacteristics H(z) to be “emulated” by the adaptive filter. The deadtime can be lumped into the delay unit 335, which allows for a moreefficient use of the adaptive filter 331.

FIG. 6 illustrates an alternative to the example of FIG. 5. In thelimiter control unit 33 of FIG. 6 the undelayed sum signal y_(SUM)[n] isfed to the filter 332 for calculating the current estimate i_(IN)[n+1]instead of the sum signal x_(SUM)[n]. Concerning this example it isimportant to notice that the above mentioned delay unit 335 (delay ofn_(delay) samples) is required and that the delayed signaly_(SUM)[n-n_(delay)] is provided to the adaptive filter 331 whereas theundelayed signal y_(SUM)[n] is provided to the filter unit 332 forcalculating the current estimate i_(IN)′[n+1]. It should be noted thatthe current estimate i_(IN)′[n+1] provided by the example of FIG. 5 isscaled by a factor G_(LIM)[n] as compared to the current estimatei_(IN)′[n+1] provided by the example of FIG. 6. However, this scalingcan be considered in the limiter gain calculation and is thusunproblematic.

FIG. 7 illustrates an enhanced example of the combining of the outputsignals y_(CH1)[n], . . . , y_(CHN)[n] to one sum signal y_(SUM)[n] asillustrated in the examples of FIGS. 5 and 6. In accordance with FIG. 7each one of the output signals y_(CH1)[n], . . . , y_(CHN)[n] is squared(or transformed to a positive definite signal) and then weighted (e.g.,multiplied) by a corresponding weighing factor c₁, c₂, . . . , c_(N).Alternatively, the weighing can be done with a factor sqrt(c_(i)) (orsqrt(c₂) etc.) before the square operation. The weighing factors c₁, c₂,c_(N) are calculated so as to compensate for loudspeakers havingdifferent impedances connected to the different audio channels. Forexample, an audio channel supplying a 4 ohm loudspeaker will have tosupply twice the output current than a channel supplying an 8 ohmloudspeaker when supplied with the same output signals y₁[], y₂[n].Consequently, the weighing factor c₁ for the first channel would equaltwo times the weighing factor for the second channel in the presentexample. As the loudspeaker's impedance does not significantly changeduring operation, the weighing factors c₁, c₂, . . . , c_(N) can beconsidered as constants. In a more sophisticated implementation theweighing factors may be frequency dependent, i.e., different weighingfactors c_(1k), c_(2k), c_(Nk) are used for different spectral ranges(spectral lines k) of the output signals y_(CHI)[n], . . . , y_(CHN)[n].Thereby, the frequency dependent weighing factors may represent thefrequency dependent impedance of the respective loudspeakers. In theexample of FIG. 5 the input signals x₁[n], x₂[n], . . . , x_(N)[] may beprocessed the same way as the output signals (see FIG. 7), so as to forma corresponding sum signal x_(SUM)[n].

For an efficient implementation the adaptive filter may be realized inthe frequency domain using fast Fourier algorithms in connection withthe known overlap-and-save method (see Oppenheim-Schafer: Chapter 8.7.3Implementing Linear Time-Invariant Systems using DFT, in: Discrete TimeSignal Processing, Prentice Hall, 1999). Further, in order to allow fora computationally more efficient implementation of the FIR filter 332(see FIG. 4) the FIR transfer function H(z) can be transformed to (andthus approximated by) a computationally more efficient infinite impulseresponse (IIR) filter using LPC analysis which can efficientlyimplemented using the Levinson-Durbin recursion. Therefore, only theminimum phase component of the estimated (FIR) transfer function H′(z)has to be considered for the transformation to a corresponding IIRfilter H″(z) as the phase response of the resulting IIR filter isirrelevant in the present application. This feature is included in thelimiter controller 33 illustrated in FIG. 8 and entails the advantage ofreduced computational complexity, as IIR filters requires significantlyless coefficients as FIR filters exhibiting a similar frequencyresponse.

The limiter 30 included in the example of FIG. 8 differs from thelimiter illustrated in the examples of FIG. 3 or 4 in that therespective limiter gain calculation unit 32 is configured to calculate afrequency dependent limiter gain G_(LIM,k)[n] whereby the subscript kdenotes the respective frequency band (or discrete “spectral line” asthe signals are processed as discrete time and discrete frequencysignals). This frequency dependent limiter gain G_(LIM,k)[n] allows toreduce the gain of some spectral components of the input signals x[n]more than the gain of other components. A reduction of the limiter gainassociated with a bass frequency range (e.g., frequencies below 180 Hz)will have a greater impact on the current consumption as a reduction ofthe limiter gain associated with the treble frequency range (e.g.,greater than 1 kHz).

In accordance with a further example of the invention the frequencydependent limiter gain G_(LIM,k)[n] exhibits a frequency characteristicsthat (at least approximately) matches the (optionally scaled) frequencycharacteristics of the inverse of frequency response H(z) estimated bythe adaptive filter 331. When, as mentioned above, using theLevinson-Durbin recursion to “simplify” the estimates FIR filter to anIIR filter, the inverse may be easily obtained by interchangingnominator and denominator of the transfer function of the IIR filterH″(z) (filter 332). However, the mentioned “simplification” of theestimated FIR filter is not necessary for implementing a frequencydependent limiter gain G_(LIM,k)[n]; the limiter gain calculation unit32 may use a different strategy to calculate the frequency dependentlimiter gain G_(LIM,k)[n]. Further, different limiter thresholdsi_(MAX,k) may be provided for the respective frequency bands (spectrallines) and typically an attenuation of the low frequencies has a greaterimpact on the current consumption than an attenuation of the higherfrequencies. When attenuating only the low frequencies, however, theperceived loudness will decrease not so much as if a broadbandattenuation would be used as in the example of FIG. 3 or 4.

In the case of multiple audio channels a sum signal y_(SUM)[n] is formedfrom all output signals y_(CH1)[n], y_(CHN)[n], y_(CHN)[n] thatcontribute to the total current consumption i_(IN)[n] of the audioamplifier. The resulting sum signal is fed into the adaptive filter 331which is a SISO system (single input single output system) in theexample of FIGS. 5 and 6. However, as an alternative to summing theoutput signals a multidimensional adaptive filter (i.e., a MISO system,multiple input single output system) may be used to estimate a transfervector representing the transfer characteristics from the output currentvector {signals y_(CH1)[n], y_(CH2)[n], . . . , y_(CHN)[n]} to theresulting input current consumption i_(IN)[n] (or alternatively theoutput current consumption). However, in practical implementations theexample of FIGS. 5 and 6 seems to be sufficient and easier to implement.

Although various examples to realize the invention have been disclosed,it will be apparent to those skilled in the art that various changes andmodifications can be made which will achieve some of the advantages ofthe invention without departing from the spirit and scope of theinvention. It will be obvious to those reasonably skilled in the artthat other components performing the same functions may be suitablysubstituted. Such modifications to the inventive concept are intended tobe covered by the appended claims Furthermore the scope of the inventionis not limited to automotive applications but may also be applied in anyother environment, e.g., in consumer applications like home cinema orthe like and also in cinema and concert halls or the like.

Although the present invention has been illustrated and described withrespect to several preferred embodiments thereof, various changes,omissions and additions to the form and detail thereof, may be made,without departing from the spirit and scope of the invention.

1. An audio amplifier comprising at least one audio channel, the audioamplifier being configured to amplify, for each audio channel, an inputaudio signal x[n] so as to provide an amplified output signal y[n], theaudio amplifier comprises: a variable gain amplifier for each of the atleast one audio channel, the variable gain amplifier receiving the inputaudio signal x[n] and providing the output signal y[n], the outputsignal y[n] corresponding to the input signal amplified by a limitergain G_(LIM)[n]; a control unit configured to estimate, based on amathematical model, an input current (i_(IN)[n]) consumption or,alternatively, a total output current of the audio amplifier thusproviding an estimated current signal (i_(IN)′[n]) due to the inputaudio signal(s) (x[n]); and a limiter gain calculation unit thatcalculates, dependent on the estimated current signal (i_(IN)′[n]), thelimiter gain (G_(LIM)[n]) such that the actual input current (i_(IN)[n])or, respectively, the total output current of the audio amplifiersubstantially does not exceed a threshold current value (i_(MAX)). 2.The audio amplifier of claim 1, further comprising a sense unitconfigured to provide a current sense signal representative of the inputcurrent consumption or the total output current of the amplifier, thecontrol unit that estimates the transfer characteristics from a signalrepresenting the amplifier input or output signal(s) to the resultingcurrent sense signal and to use the estimated transfer characteristic tocalculate the estimated current signal (i_(IN)′[n]).
 3. The audioamplifier of claim 2, where the control unit includes an adaptive filterconfigured to emulate the transfer characteristics of an a prioriunknown system which describes the relation between a signalrepresenting the output signal(s) of the amplifier output channel(s) andthe resulting input current consumption thus providing an estimatedtransfer function; and where the control unit is configured to provide acurrent estimation from the input or the output signal(s) using theestimated transfer function.
 4. The audio amplifier of claim 3, furtherincluding a delay unit coupled to the variable gain amplifier upstreamor downstream thereof
 5. The audio amplifier of one of the claim 1comprising at least two audio channels, one variable gain amplifierbeing associated with each audio channel and being configured to amplifythe respective input signal by the limiter gain; where the control unitincludes a signal combining unit configured to combine the input or theoutput signals of the audio channels so as to provide a sum signal; thecontrol unit includes an adaptive filter that is configured to estimatea transfer function describing the transfer characteristics from the sumsignal to the input current consumption or the total output current. 6.The audio amplifier of claim 5 where the control unit further includes adelay coupled upstream to the adaptive filter.
 7. The audio amplifier ofclaim 5, where the control unit further comprises a function unit thatis configured to subject the input or output audio signals to a positivedefinite transformation before being combined so as to form the sumsignal, in particular to a square operation or an absolute valueoperation.
 8. The audio amplifier of claim 1, where the limiter gain isfrequency dependent, so as to provide different gain values throughoutdifferent frequency bands or for different spectral lines.
 9. A methodfor limiting the current consumption of an audio amplifier thatcomprises at least one audio channel, an input audio signal and anoutput audio signal being associated with each audio channel, the methodcomprises: amplifying, for each audio channel, the respective inputsignal by a variable limiter gain to provide the amplified input signal;estimating, based on a mathematical model, an input current i_(IN)[n]consumption or, alternatively, a total output current of the audioamplifier thus providing an estimated current signal i_(IN)′[n]resulting from the input signal(s); calculating a limiter gain dependenton the estimated current signal i_(IN)′[n] such that the actual inputcurrent i_(IN)[n] or, respectively, the total output current of theaudio amplifier substantially does not exceed a threshold current value(i_(MAX)).
 10. The method of claim 9, where the estimating of thecurrent signal comprises: measuring a current sense signalrepresentative of the actual input current consumption or the totaloutput current of the audio amplifier; iteratively adapting finiteimpulse response (FIR) filter coefficients so as to emulate a transferfunction representing the transfer characteristics from a signalrepresenting the amplifier input or output signal(s) to the resultingcurrent sense signal; and using the estimated transfer function tocalculate the estimated current signal.
 11. The method of claim 10 wherethe step of using the estimated transfer function to calculate theestimated current signal comprises: transforming the iteratively adaptedFIR filter coefficients into corresponding infinite impulse response(IIR) filter coefficients so as to obtain an IIR filter havingapproximately the same transfer characteristics as the minimum phasepart of the FIR filter.
 12. The method of one of the claims 9 where thelimiter gain is frequency dependent.
 13. The method of claim 11 wherethe limiter gain is frequency dependent and where the step ofcalculating a limiter gain comprises: calculating an inverse transferfunction of the FIR or the IIR filter; and calculating the frequencydependent limiter gain such that the frequency characteristics of thelimiter gain at least approximately matches the frequencycharacteristics of the inverse transfer function of the FIR or the IIRfilter.
 14. The method of claim 9, where the audio amplifier comprisesat least two audio channels, the method comprises: combining the inputor the output signals of the audio channels so as to provide a sumsignal; estimating a transfer function describing the transfercharacteristics from the sum signal to the input current consumption orthe total output current.
 15. The method of claim 15 where the step ofcombining comprises a weighted summation of the input or the outputsignals of the at least two audio channels, where the weighing factorsare chosen so as to consider loudspeakers of different impedances atdifferent audio channels, the weighing factors being optionallyfrequency dependent and as such considering the individual impedancefunctions of the corresponding loudspeakers.